Grandstream UCM6200 - VoIP.ms Wiki

Grandstream UCM6200

From VoIP.ms Wiki

Jump to: navigation, search
Ucm6202.jpg

Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video conferencing, video surveillance, data tools, mobility options, and facility access management onto one common network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.

Website: UCM 6200 Series


Contents

Login into your device

  1. Connect a computer to the same network as the UCM6202
  2. Ensure the UCM 6202 is properly powered on and displays the IP address on the LCD screen
  3. Open a web browser on the computer and enter the displayed IP address into the search bar in the following format http(s)://ipaddress:portnumber. The default protocol is HTTPS and the default port number is 8089.
  4. The Web portal should be shown (see figure below). Enter the default administrator credentials:
Username: admin
Password: admin
Click to enlarge

Creating a trunk

To create a new VoIP.ms trunk head to Extension/Trunk>> VoIP trunks, from the left panel. In this section, you can choose to add a SIP or an IAX trunk.

Click to enlarge

SIP Trunk

To create a SIP trunk you only need to fill some basic information.

Note: Bear in mind to use the same VoIP server your VoIP number is using. 
You can check what VoIP server your VoIP number is using from your VoIP.ms customer portal 
at DID Numbers>> Manage DIDs and under POP column. 
You can choose any server as long as the one in your portal and the one in this field matches, 
otherwise, incoming calls won't ring.

When done, just click on the "Save" button. Your new trunk will be created but changes won't take effect until you click on the "Apply changes" button, we can do this later when the set up is complete.

If you get a pop-up window letting you know about the SIP port number, just click "Ok"

Click to enlarge

SIP settings enhanced

Once your basic SIP trunk has been created we will proceed to improve some settings on it, click on the edit icon for your trunk.

SIP headers

In order to send your SIP headers as we require, please make sure the "fromuser" field has your account or sub-account name in it. Bear in mind "username" and "fromuser" fields should contain the same information.

Click to enlarge
Audio codecs

Now click at "Advanced Settings" and at "Codec Preference" use only the supported codecs by VoIP.ms. At the right column choose only G.729, PCMU & GSM, in this order.

Trunk Caller ID

If your trunk is going to send its own CID number you can use PAI (P-Asserted-Identity) header. You only need to enable "Send PAI Header" and type the caller ID name and Caller ID number your trunk will send by using the following format: YourName<YourPhoneNumber>.

In order to send PAI header successfully, bear in mind that no CID number must be set from your VoIP.ms portal at your account or sub-accounts section, otherwise the CID number set in your portal will override the one sent by your trunk.

Keep Alive Event

We also suggest enabling "Enable Heartbeat Detection" in order to send SIP OPTIONS messages to our server and avoid your modem to close your local SIP ports. Make "Heartbeat Frequency" to 50 seconds, this should be enough time to renew your local SIP port connection.


Finally, click on the "Save" button and at this stage, you can also click on "Apply Changes". Your trunk should be shown as "Registered" from your VoIP.ms dashboard, however, no calls will work until you set up your outbound and inbound routes.

Click to enlarge
TLS

In order to use TLS along with your UCM please follow these steps:

1. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled. Bear in mind, if this setting is enabled and you use UDP/TCP you will be rejected with error code 488. Enable this for the Main Account at Main Menu>> Account settings>> Advanced tab and for a sub-account at Sub accounts>> Manage sub-accounts and by clicking on the orange icon with a pen and click at "Advanced Options Click here to display"

Click to enlarge
Click to enlarge

2. Now that your account/sub-account has this setting enabled, your device only needs to send TLS and SRTP.

Go to Extension/Trunk>> VoIP Trunks and click on "Edit trunk". In this section make sure you have the following settings:

*Host Name: toronto5.voip.ms:5061 (Use the same server your phone number is at, you can check it out from your customer portal at Manage DIDs section).

*Transport: TLS

Note: When using TLS is very important to specify the number of the server, in case the name you have chosen doesn't use the number 1 you need to add it, at least when using TLS

Click to enlarge

Go to Advanced Settings and set SRTP to "Enabled and forced"

Click to enlarge

IAX2 Trunk

To create an IAX2 trunk just fill the following information:

Note: Bear in mind to use the same VoIP server your VoIP number is using. 
You can check what VoIP server your VoIP number is using from your VoIP.ms customer portal 
at DID Numbers>> Manage DIDs and under POP column. 
You can choose any server as long as the one in your portal and the one in this field matches, 
otherwise, incoming calls won't ring.
Click to enlarge

IAX settings enhanced

Click on the edit icon for your trunk and go to "Advanced settings". Make sure only supported codecs by VoIP.ms are listed in the right column.

Also, enable "Heartbeat Frequency" to send Keep Alive events to our server and avoid closing your local ports by your modem. "Heartbeat Frequency" should be set to 50 seconds.

Click to enlarge

Creating your outbound route

Outbound routes are the ones in charge of making match your dialing pattern and send your call through the proper trunk

VoIP.ms suggest to include the following patterns into your outbound route:

_1NXXXXXXXXX
_NXXXXXXXXX
_4XXX
_00.
_011.
_033.
_044.

All your different dial patterns must be prefixed by the character "_"

To create your outbound routes click on Extension/Trunk>> Outbound routes, from the left panel and click on "Add"

Click to enlarge

In this section you only need to fill the following fields:

Click to enlarge

Note: If you want to include a dial-out prefix, you can type it after the "_" character in your dial patterns. This number will need to be stripped off, you can do this by using the "Strip" field, you can choose how many digits you can strip after the "_" character.

For example: If you want to use the "9" digit to dial out, then your pattern will need to be _9NXXXXXXXXX. 
To strip off the number "9" when dialing out you will need to set the "Strip" field to "1", this way only one character (in this case the number "9") will be stripped off.

Creating your inbound route

Thanks to your inbound routes you can use only one single trunk to receive all the incoming calls from all your phone numbers. This way you don't need to use more than one trunk for your phone numbers, inbound routes will receive all of them and route them into the proper destination in your UCM.

 Note: We do not suggest using more than one VoIP.ms trunk on the same device. 

From the left panel, head to Extension/Trunk>> Inbound Routes and click on "Add".

Click to enlarge

In this section, fill the following fields:

Click to enlarge

Note: Remember to send your phone number in the TO SIP header, this way your PBx will match it with your inbound routes. You can do this very easily by setting Device Type from "ATA adapter, IP phone or Softphone" to "IP PBX Server, Asterisk or Softswitch". You will find this setting from your VoIP.ms customer portal at Main Menu>> Account settings>> Inbound settings, if you're using the main account or at Sub accounts>> Manage Sub accounts and by clicking on the orange icon with a pen, if you're using a sub-account

Click to enlarge
Click to enlarge
Personal tools
Namespaces
Variants
Actions
VoIP.ms Wiki
Configuration
Guides (English)
Guides (Français)
Guías (Español)
Toolbox